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ladybird/Userland/Libraries/LibAudio/MP3Loader.cpp
kleines Filmröllchen 264cc76ab4 LibAudio: Move audio stream buffering into the loader 
Before, some loader plugins implemented their own buffering (FLAC&MP3),
some didn't require any (WAV), and some didn't buffer at all (QOA). This
meant that in practice, while you could load arbitrary amounts of
samples from some loader plugins, you couldn't do that with some others.
Also, it was ill-defined how many samples you would actually get back
from a get_more_samples call.

This commit fixes that by introducing a layer of abstraction between the
loader and its plugins (because that's the whole point of having the
extra class!). The plugins now only implement a load_chunks() function,
which is much simpler to implement and allows plugins to play fast and
loose with what they actually return. Basically, they can return many
chunks of samples, where one chunk is simply a convenient block of
samples to load. In fact, some loaders such as FLAC and QOA have
separate internal functions for loading exactly one chunk. The loaders
*should* load as many chunks as necessary for the sample count to be
reached or surpassed (the latter simplifies loading loops in the
implementations, since you don't need to know how large your next chunk
is going to be; a problem for e.g. FLAC). If a plugin has no problems
returning data of arbitrary size (currently WAV), it can return a single
chunk that exactly (or roughly) matches the requested sample count. If a
plugin is at the stream end, it can also return less samples than was
requested! The loader can handle all of these cases and may call into
load_chunk multiple times. If the plugin returns an empty chunk list (or
only empty chunks; again, they can play fast and loose), the loader
takes that as a stream end signal. Otherwise, the loader will always
return exactly as many samples as the user requested. Buffering is
handled by the loader, allowing any underlying plugin to deal with any
weird sample count requirement the user throws at it (looking at you,
SoundPlayer!).

This (not accidentally!) makes QOA work in SoundPlayer.
2023-03-13 13:25:42 +01:00

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/*
* Copyright (c) 2021, Arne Elster <arne@elster.li>
*
* SPDX-License-Identifier: BSD-2-Clause
*/
#include "MP3Loader.h"
#include "MP3HuffmanTables.h"
#include "MP3Tables.h"
#include <AK/FixedArray.h>
#include <LibCore/File.h>
namespace Audio {
DSP::MDCT<12> MP3LoaderPlugin::s_mdct_12;
DSP::MDCT<36> MP3LoaderPlugin::s_mdct_36;
MP3LoaderPlugin::MP3LoaderPlugin(NonnullOwnPtr<SeekableStream> stream)
: LoaderPlugin(move(stream))
{
}
Result<NonnullOwnPtr<MP3LoaderPlugin>, LoaderError> MP3LoaderPlugin::create(StringView path)
{
auto stream = LOADER_TRY(Core::BufferedFile::create(LOADER_TRY(Core::File::open(path, Core::File::OpenMode::Read))));
auto loader = make<MP3LoaderPlugin>(move(stream));
LOADER_TRY(loader->initialize());
return loader;
}
Result<NonnullOwnPtr<MP3LoaderPlugin>, LoaderError> MP3LoaderPlugin::create(Bytes buffer)
{
auto stream = LOADER_TRY(try_make<FixedMemoryStream>(buffer));
auto loader = make<MP3LoaderPlugin>(move(stream));
LOADER_TRY(loader->initialize());
return loader;
}
MaybeLoaderError MP3LoaderPlugin::initialize()
{
m_bitstream = LOADER_TRY(try_make<BigEndianInputBitStream>(MaybeOwned<Stream>(*m_stream)));
TRY(synchronize());
auto header = TRY(read_header());
if (header.id != 1 || header.layer != 3)
return LoaderError { LoaderError::Category::Format, "Only MPEG-1 layer 3 supported." };
m_sample_rate = header.samplerate;
m_num_channels = header.channel_count();
m_loaded_samples = 0;
TRY(build_seek_table());
LOADER_TRY(m_stream->seek(0, SeekMode::SetPosition));
return {};
}
MaybeLoaderError MP3LoaderPlugin::reset()
{
TRY(seek(0));
m_current_frame = {};
m_synthesis_buffer = {};
m_loaded_samples = 0;
LOADER_TRY(m_bit_reservoir.discard(m_bit_reservoir.used_buffer_size()));
m_bitstream->align_to_byte_boundary();
return {};
}
MaybeLoaderError MP3LoaderPlugin::seek(int const position)
{
for (auto const& seek_entry : m_seek_table) {
if (seek_entry.get<1>() >= position) {
LOADER_TRY(m_stream->seek(seek_entry.get<0>(), SeekMode::SetPosition));
m_loaded_samples = seek_entry.get<1>();
break;
}
}
m_current_frame = {};
m_synthesis_buffer = {};
LOADER_TRY(m_bit_reservoir.discard(m_bit_reservoir.used_buffer_size()));
m_bitstream->align_to_byte_boundary();
return {};
}
ErrorOr<Vector<FixedArray<Sample>>, LoaderError> MP3LoaderPlugin::load_chunks(size_t samples_to_read_from_input)
{
int samples_to_read = samples_to_read_from_input;
Vector<FixedArray<Sample>> frames;
while (samples_to_read > 0) {
FixedArray<Sample> samples = LOADER_TRY(FixedArray<Sample>::create(1152));
if (!m_current_frame.has_value()) {
auto maybe_frame = read_next_frame();
if (maybe_frame.is_error()) {
if (m_stream->is_eof()) {
return Vector<FixedArray<Sample>> {};
}
return maybe_frame.release_error();
}
m_current_frame = maybe_frame.release_value();
if (!m_current_frame.has_value())
break;
}
bool const is_stereo = m_current_frame->header.channel_count() == 2;
auto current_frame_read = 0;
for (; current_frame_read < 576; current_frame_read++) {
auto const left_sample = m_current_frame->channels[0].granules[0].pcm[current_frame_read / 32][current_frame_read % 32];
auto const right_sample = is_stereo ? m_current_frame->channels[1].granules[0].pcm[current_frame_read / 32][current_frame_read % 32] : left_sample;
samples[current_frame_read] = Sample { left_sample, right_sample };
samples_to_read--;
}
for (; current_frame_read < 1152; current_frame_read++) {
auto const left_sample = m_current_frame->channels[0].granules[1].pcm[(current_frame_read - 576) / 32][(current_frame_read - 576) % 32];
auto const right_sample = is_stereo ? m_current_frame->channels[1].granules[1].pcm[(current_frame_read - 576) / 32][(current_frame_read - 576) % 32] : left_sample;
samples[current_frame_read] = Sample { left_sample, right_sample };
samples_to_read--;
}
m_loaded_samples += samples.size();
TRY(frames.try_append(move(samples)));
m_current_frame = {};
}
return frames;
}
MaybeLoaderError MP3LoaderPlugin::build_seek_table()
{
int sample_count = 0;
size_t frame_count = 0;
m_seek_table.clear();
m_bitstream->align_to_byte_boundary();
while (!synchronize().is_error()) {
auto const frame_pos = -2 + LOADER_TRY(m_stream->seek(0, SeekMode::FromCurrentPosition));
auto error_or_header = read_header();
if (error_or_header.is_error() || error_or_header.value().id != 1 || error_or_header.value().layer != 3) {
continue;
}
frame_count++;
sample_count += 1152;
if (frame_count % 10 == 0)
m_seek_table.append({ frame_pos, sample_count });
LOADER_TRY(m_stream->seek(error_or_header.value().frame_size - 6, SeekMode::FromCurrentPosition));
// TODO: This is just here to clear the bitstream buffer.
// Bitstream should have a method to sync its state to the underlying stream.
m_bitstream->align_to_byte_boundary();
}
m_total_samples = sample_count;
return {};
}
ErrorOr<MP3::Header, LoaderError> MP3LoaderPlugin::read_header()
{
MP3::Header header;
header.id = LOADER_TRY(m_bitstream->read_bit());
header.layer = MP3::Tables::LayerNumberLookup[LOADER_TRY(m_bitstream->read_bits(2))];
if (header.layer <= 0)
return LoaderError { LoaderError::Category::Format, m_loaded_samples, "Frame header contains invalid layer number." };
header.protection_bit = LOADER_TRY(m_bitstream->read_bit());
header.bitrate = MP3::Tables::BitratesPerLayerLookup[header.layer - 1][LOADER_TRY(m_bitstream->read_bits(4))];
if (header.bitrate <= 0)
return LoaderError { LoaderError::Category::Format, m_loaded_samples, "Frame header contains invalid bitrate." };
header.samplerate = MP3::Tables::SampleratesLookup[LOADER_TRY(m_bitstream->read_bits(2))];
if (header.samplerate <= 0)
return LoaderError { LoaderError::Category::Format, m_loaded_samples, "Frame header contains invalid samplerate." };
header.padding_bit = LOADER_TRY(m_bitstream->read_bit());
header.private_bit = LOADER_TRY(m_bitstream->read_bit());
header.mode = static_cast<MP3::Mode>(LOADER_TRY(m_bitstream->read_bits(2)));
header.mode_extension = static_cast<MP3::ModeExtension>(LOADER_TRY(m_bitstream->read_bits(2)));
header.copyright_bit = LOADER_TRY(m_bitstream->read_bit());
header.original_bit = LOADER_TRY(m_bitstream->read_bit());
header.emphasis = static_cast<MP3::Emphasis>(LOADER_TRY(m_bitstream->read_bits(2)));
if (!header.protection_bit)
header.crc16 = LOADER_TRY(m_bitstream->read_bits<u16>(16));
header.frame_size = 144 * header.bitrate * 1000 / header.samplerate + header.padding_bit;
header.slot_count = header.frame_size - ((header.channel_count() == 2 ? 32 : 17) + (header.protection_bit ? 0 : 2) + 4);
return header;
}
MaybeLoaderError MP3LoaderPlugin::synchronize()
{
size_t one_counter = 0;
while (one_counter < 12 && !m_bitstream->is_eof()) {
bool const bit = LOADER_TRY(m_bitstream->read_bit());
one_counter = bit ? one_counter + 1 : 0;
if (!bit) {
m_bitstream->align_to_byte_boundary();
}
}
if (one_counter != 12)
return LoaderError { LoaderError::Category::Format, m_loaded_samples, "Failed to synchronize." };
return {};
}
ErrorOr<MP3::MP3Frame, LoaderError> MP3LoaderPlugin::read_next_frame()
{
// Note: This will spin until we find a correct frame, or we reach eof.
// In the second case, the error will bubble up from read_frame_data().
while (true) {
TRY(synchronize());
MP3::Header header = TRY(read_header());
if (header.id != 1 || header.layer != 3) {
continue;
}
return read_frame_data(header);
}
}
ErrorOr<MP3::MP3Frame, LoaderError> MP3LoaderPlugin::read_frame_data(MP3::Header const& header)
{
MP3::MP3Frame frame { header };
TRY(read_side_information(frame));
auto maybe_buffer = ByteBuffer::create_uninitialized(header.slot_count);
if (maybe_buffer.is_error())
return LoaderError { LoaderError::Category::IO, m_loaded_samples, "Out of memory" };
auto& buffer = maybe_buffer.value();
size_t old_reservoir_size = m_bit_reservoir.used_buffer_size();
LOADER_TRY(m_bitstream->read_entire_buffer(buffer));
if (LOADER_TRY(m_bit_reservoir.write(buffer)) != header.slot_count)
return LoaderError { LoaderError::Category::IO, m_loaded_samples, "Could not write frame into bit reservoir." };
// If we don't have enough data in the reservoir to process this frame, skip it (but keep the data).
if (old_reservoir_size < static_cast<size_t>(frame.main_data_begin))
return frame;
TRY(m_bit_reservoir.discard(old_reservoir_size - frame.main_data_begin));
BigEndianInputBitStream reservoir_stream { MaybeOwned<Stream>(m_bit_reservoir) };
for (size_t granule_index = 0; granule_index < 2; granule_index++) {
for (size_t channel_index = 0; channel_index < header.channel_count(); channel_index++) {
size_t scale_factor_size = TRY(read_scale_factors(frame, reservoir_stream, granule_index, channel_index));
TRY(read_huffman_data(frame, reservoir_stream, granule_index, channel_index, scale_factor_size));
if (frame.channels[channel_index].granules[granule_index].block_type == MP3::BlockType::Short) {
reorder_samples(frame.channels[channel_index].granules[granule_index], frame.header.samplerate);
// Only reduce alias for lowest 2 bands as they're long.
// Afaik this is not mentioned in the ISO spec, but it is addressed in the
// changelog for the ISO compliance tests.
if (frame.channels[channel_index].granules[granule_index].mixed_block_flag)
reduce_alias(frame.channels[channel_index].granules[granule_index], 36);
} else {
reduce_alias(frame.channels[channel_index].granules[granule_index]);
}
}
if (header.mode == MP3::Mode::JointStereo) {
process_stereo(frame, granule_index);
}
}
for (size_t granule_index = 0; granule_index < 2; granule_index++) {
for (size_t channel_index = 0; channel_index < header.channel_count(); channel_index++) {
auto& granule = frame.channels[channel_index].granules[granule_index];
for (size_t i = 0; i < 576; i += 18) {
MP3::BlockType block_type = granule.block_type;
if (i < 36 && granule.mixed_block_flag) {
// ISO/IEC 11172-3: if mixed_block_flag is set, the lowest two subbands are transformed with normal window.
block_type = MP3::BlockType::Normal;
}
Array<float, 36> output;
transform_samples_to_time(granule.samples, i, output, block_type);
int const subband_index = i / 18;
for (size_t sample_index = 0; sample_index < 18; sample_index++) {
// overlap add
granule.filter_bank_input[subband_index][sample_index] = output[sample_index] + m_last_values[channel_index][subband_index][sample_index];
m_last_values[channel_index][subband_index][sample_index] = output[sample_index + 18];
// frequency inversion
if (subband_index % 2 == 1 && sample_index % 2 == 1)
granule.filter_bank_input[subband_index][sample_index] *= -1;
}
}
}
}
Array<float, 32> in_samples;
for (size_t channel_index = 0; channel_index < frame.header.channel_count(); channel_index++) {
for (size_t granule_index = 0; granule_index < 2; granule_index++) {
auto& granule = frame.channels[channel_index].granules[granule_index];
for (size_t sample_index = 0; sample_index < 18; sample_index++) {
for (size_t band_index = 0; band_index < 32; band_index++) {
in_samples[band_index] = granule.filter_bank_input[band_index][sample_index];
}
synthesis(m_synthesis_buffer[channel_index], in_samples, granule.pcm[sample_index]);
}
}
}
return frame;
}
MaybeLoaderError MP3LoaderPlugin::read_side_information(MP3::MP3Frame& frame)
{
frame.main_data_begin = LOADER_TRY(m_bitstream->read_bits(9));
if (frame.header.channel_count() == 1) {
frame.private_bits = LOADER_TRY(m_bitstream->read_bits(5));
} else {
frame.private_bits = LOADER_TRY(m_bitstream->read_bits(3));
}
for (size_t channel_index = 0; channel_index < frame.header.channel_count(); channel_index++) {
for (size_t scale_factor_selection_info_band = 0; scale_factor_selection_info_band < 4; scale_factor_selection_info_band++) {
frame.channels[channel_index].scale_factor_selection_info[scale_factor_selection_info_band] = LOADER_TRY(m_bitstream->read_bit());
}
}
for (size_t granule_index = 0; granule_index < 2; granule_index++) {
for (size_t channel_index = 0; channel_index < frame.header.channel_count(); channel_index++) {
auto& granule = frame.channels[channel_index].granules[granule_index];
granule.part_2_3_length = LOADER_TRY(m_bitstream->read_bits(12));
granule.big_values = LOADER_TRY(m_bitstream->read_bits(9));
granule.global_gain = LOADER_TRY(m_bitstream->read_bits(8));
granule.scalefac_compress = LOADER_TRY(m_bitstream->read_bits(4));
granule.window_switching_flag = LOADER_TRY(m_bitstream->read_bit());
if (granule.window_switching_flag) {
granule.block_type = static_cast<MP3::BlockType>(LOADER_TRY(m_bitstream->read_bits(2)));
granule.mixed_block_flag = LOADER_TRY(m_bitstream->read_bit());
for (size_t region = 0; region < 2; region++)
granule.table_select[region] = LOADER_TRY(m_bitstream->read_bits(5));
for (size_t window = 0; window < 3; window++)
granule.sub_block_gain[window] = LOADER_TRY(m_bitstream->read_bits(3));
granule.region0_count = (granule.block_type == MP3::BlockType::Short && !granule.mixed_block_flag) ? 8 : 7;
granule.region1_count = 36;
} else {
for (size_t region = 0; region < 3; region++)
granule.table_select[region] = LOADER_TRY(m_bitstream->read_bits(5));
granule.region0_count = LOADER_TRY(m_bitstream->read_bits(4));
granule.region1_count = LOADER_TRY(m_bitstream->read_bits(3));
}
granule.preflag = LOADER_TRY(m_bitstream->read_bit());
granule.scalefac_scale = LOADER_TRY(m_bitstream->read_bit());
granule.count1table_select = LOADER_TRY(m_bitstream->read_bit());
}
}
return {};
}
// From ISO/IEC 11172-3 (2.4.3.4.7.1)
Array<float, 576> MP3LoaderPlugin::calculate_frame_exponents(MP3::MP3Frame const& frame, size_t granule_index, size_t channel_index)
{
Array<float, 576> exponents;
auto fill_band = [&exponents](float exponent, size_t start, size_t end) {
for (size_t j = start; j <= end; j++) {
exponents[j] = exponent;
}
};
auto const& channel = frame.channels[channel_index];
auto const& granule = frame.channels[channel_index].granules[granule_index];
auto const scale_factor_bands = get_scalefactor_bands(granule, frame.header.samplerate);
float const scale_factor_multiplier = granule.scalefac_scale ? 1 : 0.5;
int const gain = granule.global_gain - 210;
if (granule.block_type != MP3::BlockType::Short) {
for (size_t band_index = 0; band_index < 22; band_index++) {
float const exponent = gain / 4.0f - (scale_factor_multiplier * (channel.scale_factors[band_index] + granule.preflag * MP3::Tables::Pretab[band_index]));
fill_band(AK::pow<float>(2.0, exponent), scale_factor_bands[band_index].start, scale_factor_bands[band_index].end);
}
} else {
size_t band_index = 0;
size_t sample_count = 0;
if (granule.mixed_block_flag) {
while (sample_count < 36) {
float const exponent = gain / 4.0f - (scale_factor_multiplier * (channel.scale_factors[band_index] + granule.preflag * MP3::Tables::Pretab[band_index]));
fill_band(AK::pow<float>(2.0, exponent), scale_factor_bands[band_index].start, scale_factor_bands[band_index].end);
sample_count += scale_factor_bands[band_index].width;
band_index++;
}
}
float const gain0 = (gain - 8 * granule.sub_block_gain[0]) / 4.0;
float const gain1 = (gain - 8 * granule.sub_block_gain[1]) / 4.0;
float const gain2 = (gain - 8 * granule.sub_block_gain[2]) / 4.0;
while (sample_count < 576 && band_index < scale_factor_bands.size()) {
float const exponent0 = gain0 - (scale_factor_multiplier * channel.scale_factors[band_index + 0]);
float const exponent1 = gain1 - (scale_factor_multiplier * channel.scale_factors[band_index + 1]);
float const exponent2 = gain2 - (scale_factor_multiplier * channel.scale_factors[band_index + 2]);
fill_band(AK::pow<float>(2.0, exponent0), scale_factor_bands[band_index + 0].start, scale_factor_bands[band_index + 0].end);
sample_count += scale_factor_bands[band_index + 0].width;
fill_band(AK::pow<float>(2.0, exponent1), scale_factor_bands[band_index + 1].start, scale_factor_bands[band_index + 1].end);
sample_count += scale_factor_bands[band_index + 1].width;
fill_band(AK::pow<float>(2.0, exponent2), scale_factor_bands[band_index + 2].start, scale_factor_bands[band_index + 2].end);
sample_count += scale_factor_bands[band_index + 2].width;
band_index += 3;
}
while (sample_count < 576)
exponents[sample_count++] = 0;
}
return exponents;
}
ErrorOr<size_t, LoaderError> MP3LoaderPlugin::read_scale_factors(MP3::MP3Frame& frame, BigEndianInputBitStream& reservoir, size_t granule_index, size_t channel_index)
{
auto& channel = frame.channels[channel_index];
auto const& granule = channel.granules[granule_index];
size_t band_index = 0;
size_t bits_read = 0;
if (granule.window_switching_flag && granule.block_type == MP3::BlockType::Short) {
if (granule.mixed_block_flag) {
for (size_t i = 0; i < 8; i++) {
auto const bits = MP3::Tables::ScalefacCompressSlen1[granule.scalefac_compress];
channel.scale_factors[band_index++] = TRY(reservoir.read_bits(bits));
bits_read += bits;
}
for (size_t i = 3; i < 12; i++) {
auto const bits = i <= 5 ? MP3::Tables::ScalefacCompressSlen1[granule.scalefac_compress] : MP3::Tables::ScalefacCompressSlen2[granule.scalefac_compress];
channel.scale_factors[band_index++] = TRY(reservoir.read_bits(bits));
channel.scale_factors[band_index++] = TRY(reservoir.read_bits(bits));
channel.scale_factors[band_index++] = TRY(reservoir.read_bits(bits));
bits_read += 3 * bits;
}
} else {
for (size_t i = 0; i < 12; i++) {
auto const bits = i <= 5 ? MP3::Tables::ScalefacCompressSlen1[granule.scalefac_compress] : MP3::Tables::ScalefacCompressSlen2[granule.scalefac_compress];
channel.scale_factors[band_index++] = TRY(reservoir.read_bits(bits));
channel.scale_factors[band_index++] = TRY(reservoir.read_bits(bits));
channel.scale_factors[band_index++] = TRY(reservoir.read_bits(bits));
bits_read += 3 * bits;
}
}
channel.scale_factors[band_index++] = 0;
channel.scale_factors[band_index++] = 0;
channel.scale_factors[band_index++] = 0;
} else {
if ((channel.scale_factor_selection_info[0] == 0) || (granule_index == 0)) {
for (band_index = 0; band_index < 6; band_index++) {
auto const bits = MP3::Tables::ScalefacCompressSlen1[granule.scalefac_compress];
channel.scale_factors[band_index] = TRY(reservoir.read_bits(bits));
bits_read += bits;
}
}
if ((channel.scale_factor_selection_info[1] == 0) || (granule_index == 0)) {
for (band_index = 6; band_index < 11; band_index++) {
auto const bits = MP3::Tables::ScalefacCompressSlen1[granule.scalefac_compress];
channel.scale_factors[band_index] = TRY(reservoir.read_bits(bits));
bits_read += bits;
}
}
if ((channel.scale_factor_selection_info[2] == 0) || (granule_index == 0)) {
for (band_index = 11; band_index < 16; band_index++) {
auto const bits = MP3::Tables::ScalefacCompressSlen2[granule.scalefac_compress];
channel.scale_factors[band_index] = TRY(reservoir.read_bits(bits));
bits_read += bits;
}
}
if ((channel.scale_factor_selection_info[3] == 0) || (granule_index == 0)) {
for (band_index = 16; band_index < 21; band_index++) {
auto const bits = MP3::Tables::ScalefacCompressSlen2[granule.scalefac_compress];
channel.scale_factors[band_index] = TRY(reservoir.read_bits(bits));
bits_read += bits;
}
}
channel.scale_factors[21] = 0;
}
return bits_read;
}
MaybeLoaderError MP3LoaderPlugin::read_huffman_data(MP3::MP3Frame& frame, BigEndianInputBitStream& reservoir, size_t granule_index, size_t channel_index, size_t granule_bits_read)
{
auto const exponents = calculate_frame_exponents(frame, granule_index, channel_index);
auto& granule = frame.channels[channel_index].granules[granule_index];
auto const scale_factor_bands = get_scalefactor_bands(granule, frame.header.samplerate);
size_t const scale_factor_band_index1 = granule.region0_count + 1;
size_t const scale_factor_band_index2 = min(scale_factor_bands.size() - 1, scale_factor_band_index1 + granule.region1_count + 1);
bool const is_short_granule = granule.window_switching_flag && granule.block_type == MP3::BlockType::Short;
size_t const region1_start = is_short_granule ? 36 : scale_factor_bands[scale_factor_band_index1].start;
size_t const region2_start = is_short_granule ? 576 : scale_factor_bands[scale_factor_band_index2].start;
auto requantize = [](int const sample, float const exponent) -> float {
int const sign = sample < 0 ? -1 : 1;
int const magnitude = AK::abs(sample);
return sign * AK::pow<float>(static_cast<float>(magnitude), 4 / 3.0) * exponent;
};
size_t count = 0;
for (; count < granule.big_values * 2; count += 2) {
MP3::Tables::Huffman::HuffmanTreeXY const* tree = nullptr;
if (count < region1_start) {
tree = &MP3::Tables::Huffman::HuffmanTreesXY[granule.table_select[0]];
} else if (count < region2_start) {
tree = &MP3::Tables::Huffman::HuffmanTreesXY[granule.table_select[1]];
} else {
tree = &MP3::Tables::Huffman::HuffmanTreesXY[granule.table_select[2]];
}
if (!tree || tree->nodes.is_empty()) {
return LoaderError { LoaderError::Category::Format, m_loaded_samples, "Frame references invalid huffman table." };
}
// Assumption: There's enough bits to read. 32 is just a placeholder for "unlimited".
// There are no 32 bit long huffman codes in the tables.
auto const entry = MP3::Tables::Huffman::huffman_decode(reservoir, tree->nodes, 32);
granule_bits_read += entry.bits_read;
if (!entry.code.has_value())
return LoaderError { LoaderError::Category::Format, m_loaded_samples, "Frame contains invalid huffman data." };
int x = entry.code->symbol.x;
int y = entry.code->symbol.y;
if (x == 15 && tree->linbits > 0) {
x += LOADER_TRY(reservoir.read_bits(tree->linbits));
granule_bits_read += tree->linbits;
}
if (x != 0) {
if (LOADER_TRY(reservoir.read_bit()))
x = -x;
granule_bits_read++;
}
if (y == 15 && tree->linbits > 0) {
y += LOADER_TRY(reservoir.read_bits(tree->linbits));
granule_bits_read += tree->linbits;
}
if (y != 0) {
if (LOADER_TRY(reservoir.read_bit()))
y = -y;
granule_bits_read++;
}
granule.samples[count + 0] = requantize(x, exponents[count + 0]);
granule.samples[count + 1] = requantize(y, exponents[count + 1]);
}
ReadonlySpan<MP3::Tables::Huffman::HuffmanNode<MP3::Tables::Huffman::HuffmanVWXY>> count1table = granule.count1table_select ? MP3::Tables::Huffman::TreeB : MP3::Tables::Huffman::TreeA;
// count1 is not known. We have to read huffman encoded values
// until we've exhausted the granule's bits. We know the size of
// the granule from part2_3_length, which is the number of bits
// used for scaleactors and huffman data (in the granule).
while (granule_bits_read < granule.part_2_3_length && count <= 576 - 4) {
auto const entry = MP3::Tables::Huffman::huffman_decode(reservoir, count1table, granule.part_2_3_length - granule_bits_read);
granule_bits_read += entry.bits_read;
if (!entry.code.has_value())
return LoaderError { LoaderError::Category::Format, m_loaded_samples, "Frame contains invalid huffman data." };
int v = entry.code->symbol.v;
if (v != 0) {
if (granule_bits_read >= granule.part_2_3_length)
break;
if (LOADER_TRY(reservoir.read_bit()))
v = -v;
granule_bits_read++;
}
int w = entry.code->symbol.w;
if (w != 0) {
if (granule_bits_read >= granule.part_2_3_length)
break;
if (LOADER_TRY(reservoir.read_bit()))
w = -w;
granule_bits_read++;
}
int x = entry.code->symbol.x;
if (x != 0) {
if (granule_bits_read >= granule.part_2_3_length)
break;
if (LOADER_TRY(reservoir.read_bit()))
x = -x;
granule_bits_read++;
}
int y = entry.code->symbol.y;
if (y != 0) {
if (granule_bits_read >= granule.part_2_3_length)
break;
if (LOADER_TRY(reservoir.read_bit()))
y = -y;
granule_bits_read++;
}
granule.samples[count + 0] = requantize(v, exponents[count + 0]);
granule.samples[count + 1] = requantize(w, exponents[count + 1]);
granule.samples[count + 2] = requantize(x, exponents[count + 2]);
granule.samples[count + 3] = requantize(y, exponents[count + 3]);
count += 4;
}
if (granule_bits_read > granule.part_2_3_length) {
return LoaderError { LoaderError::Category::Format, m_loaded_samples, "Read too many bits from bit reservoir." };
}
for (size_t i = granule_bits_read; i < granule.part_2_3_length; i++) {
LOADER_TRY(reservoir.read_bit());
}
return {};
}
void MP3LoaderPlugin::reorder_samples(MP3::Granule& granule, u32 sample_rate)
{
float tmp[576] = {};
size_t band_index = 0;
size_t subband_index = 0;
auto scale_factor_bands = get_scalefactor_bands(granule, sample_rate);
if (granule.mixed_block_flag) {
while (subband_index < 36) {
for (size_t frequency_line_index = 0; frequency_line_index < scale_factor_bands[band_index].width; frequency_line_index++) {
tmp[subband_index] = granule.samples[subband_index];
subband_index++;
}
band_index++;
}
}
while (subband_index < 576 && band_index <= 36) {
for (size_t frequency_line_index = 0; frequency_line_index < scale_factor_bands[band_index].width; frequency_line_index++) {
tmp[subband_index++] = granule.samples[scale_factor_bands[band_index + 0].start + frequency_line_index];
tmp[subband_index++] = granule.samples[scale_factor_bands[band_index + 1].start + frequency_line_index];
tmp[subband_index++] = granule.samples[scale_factor_bands[band_index + 2].start + frequency_line_index];
}
band_index += 3;
}
for (size_t i = 0; i < 576; i++)
granule.samples[i] = tmp[i];
}
void MP3LoaderPlugin::reduce_alias(MP3::Granule& granule, size_t max_subband_index)
{
for (size_t subband = 0; subband < max_subband_index - 18; subband += 18) {
for (size_t i = 0; i < 8; i++) {
size_t const idx1 = subband + 17 - i;
size_t const idx2 = subband + 18 + i;
auto const d1 = granule.samples[idx1];
auto const d2 = granule.samples[idx2];
granule.samples[idx1] = d1 * MP3::Tables::AliasReductionCs[i] - d2 * MP3::Tables::AliasReductionCa[i];
granule.samples[idx2] = d2 * MP3::Tables::AliasReductionCs[i] + d1 * MP3::Tables::AliasReductionCa[i];
}
}
}
void MP3LoaderPlugin::process_stereo(MP3::MP3Frame& frame, size_t granule_index)
{
size_t band_index_ms_start = 0;
size_t band_index_ms_end = 0;
size_t band_index_intensity_start = 0;
size_t band_index_intensity_end = 0;
auto& granule_left = frame.channels[0].granules[granule_index];
auto& granule_right = frame.channels[1].granules[granule_index];
auto get_last_nonempty_band = [](Span<float> samples, ReadonlySpan<MP3::Tables::ScaleFactorBand> bands) -> size_t {
size_t last_nonempty_band = 0;
for (size_t i = 0; i < bands.size(); i++) {
bool is_empty = true;
for (size_t l = bands[i].start; l < bands[i].end; l++) {
if (samples[l] != 0) {
is_empty = false;
break;
}
}
if (!is_empty)
last_nonempty_band = i;
}
return last_nonempty_band;
};
auto process_ms_stereo = [&](MP3::Tables::ScaleFactorBand const& band) {
float const SQRT_2 = AK::sqrt(2.0);
for (size_t i = band.start; i <= band.end; i++) {
float const m = granule_left.samples[i];
float const s = granule_right.samples[i];
granule_left.samples[i] = (m + s) / SQRT_2;
granule_right.samples[i] = (m - s) / SQRT_2;
}
};
auto process_intensity_stereo = [&](MP3::Tables::ScaleFactorBand const& band, float intensity_stereo_ratio) {
for (size_t i = band.start; i <= band.end; i++) {
float const sample_left = granule_left.samples[i];
float const coeff_l = intensity_stereo_ratio / (1 + intensity_stereo_ratio);
float const coeff_r = 1 / (1 + intensity_stereo_ratio);
granule_left.samples[i] = sample_left * coeff_l;
granule_right.samples[i] = sample_left * coeff_r;
}
};
auto scale_factor_bands = get_scalefactor_bands(granule_right, frame.header.samplerate);
if (has_flag(frame.header.mode_extension, MP3::ModeExtension::MsStereo)) {
band_index_ms_start = 0;
band_index_ms_end = scale_factor_bands.size();
}
if (has_flag(frame.header.mode_extension, MP3::ModeExtension::IntensityStereo)) {
band_index_intensity_start = get_last_nonempty_band(granule_right.samples, scale_factor_bands);
band_index_intensity_end = scale_factor_bands.size();
band_index_ms_end = band_index_intensity_start;
}
for (size_t band_index = band_index_ms_start; band_index < band_index_ms_end; band_index++) {
process_ms_stereo(scale_factor_bands[band_index]);
}
for (size_t band_index = band_index_intensity_start; band_index < band_index_intensity_end; band_index++) {
auto const intensity_stereo_position = frame.channels[1].scale_factors[band_index];
if (intensity_stereo_position == 7) {
if (has_flag(frame.header.mode_extension, MP3::ModeExtension::MsStereo))
process_ms_stereo(scale_factor_bands[band_index]);
continue;
}
float const intensity_stereo_ratio = AK::tan(intensity_stereo_position * AK::Pi<float> / 12);
process_intensity_stereo(scale_factor_bands[band_index], intensity_stereo_ratio);
}
}
void MP3LoaderPlugin::transform_samples_to_time(Array<float, 576> const& input, size_t input_offset, Array<float, 36>& output, MP3::BlockType block_type)
{
if (block_type == MP3::BlockType::Short) {
size_t const N = 12;
Array<float, N * 3> temp_out;
Array<float, N / 2> temp_in;
for (size_t k = 0; k < N / 2; k++)
temp_in[k] = input[input_offset + 3 * k + 0];
s_mdct_12.transform(temp_in, Span<float>(temp_out).slice(0, N));
for (size_t i = 0; i < N; i++)
temp_out[i + 0] *= MP3::Tables::WindowBlockTypeShort[i];
for (size_t k = 0; k < N / 2; k++)
temp_in[k] = input[input_offset + 3 * k + 1];
s_mdct_12.transform(temp_in, Span<float>(temp_out).slice(12, N));
for (size_t i = 0; i < N; i++)
temp_out[i + 12] *= MP3::Tables::WindowBlockTypeShort[i];
for (size_t k = 0; k < N / 2; k++)
temp_in[k] = input[input_offset + 3 * k + 2];
s_mdct_12.transform(temp_in, Span<float>(temp_out).slice(24, N));
for (size_t i = 0; i < N; i++)
temp_out[i + 24] *= MP3::Tables::WindowBlockTypeShort[i];
Span<float> idmct1 = Span<float>(temp_out).slice(0, 12);
Span<float> idmct2 = Span<float>(temp_out).slice(12, 12);
Span<float> idmct3 = Span<float>(temp_out).slice(24, 12);
for (size_t i = 0; i < 6; i++)
output[i] = 0;
for (size_t i = 6; i < 12; i++)
output[i] = idmct1[i - 6];
for (size_t i = 12; i < 18; i++)
output[i] = idmct1[i - 6] + idmct2[i - 12];
for (size_t i = 18; i < 24; i++)
output[i] = idmct2[i - 12] + idmct3[i - 18];
for (size_t i = 24; i < 30; i++)
output[i] = idmct3[i - 18];
for (size_t i = 30; i < 36; i++)
output[i] = 0;
} else {
s_mdct_36.transform(ReadonlySpan<float>(input).slice(input_offset, 18), output);
for (size_t i = 0; i < 36; i++) {
switch (block_type) {
case MP3::BlockType::Normal:
output[i] *= MP3::Tables::WindowBlockTypeNormal[i];
break;
case MP3::BlockType::Start:
output[i] *= MP3::Tables::WindowBlockTypeStart[i];
break;
case MP3::BlockType::End:
output[i] *= MP3::Tables::WindowBlockTypeEnd[i];
break;
case MP3::BlockType::Short:
VERIFY_NOT_REACHED();
break;
}
}
}
}
// ISO/IEC 11172-3 (Figure A.2)
void MP3LoaderPlugin::synthesis(Array<float, 1024>& V, Array<float, 32>& samples, Array<float, 32>& result)
{
for (size_t i = 1023; i >= 64; i--) {
V[i] = V[i - 64];
}
for (size_t i = 0; i < 64; i++) {
V[i] = 0;
for (size_t k = 0; k < 32; k++) {
float const N = MP3::Tables::SynthesisSubbandFilterCoefficients[i][k];
V[i] += N * samples[k];
}
}
Array<float, 512> U;
for (size_t i = 0; i < 8; i++) {
for (size_t j = 0; j < 32; j++) {
U[i * 64 + j] = V[i * 128 + j];
U[i * 64 + 32 + j] = V[i * 128 + 96 + j];
}
}
Array<float, 512> W;
for (size_t i = 0; i < 512; i++) {
W[i] = U[i] * MP3::Tables::WindowSynthesis[i];
}
for (size_t j = 0; j < 32; j++) {
result[j] = 0;
for (size_t k = 0; k < 16; k++) {
result[j] += W[j + 32 * k];
}
}
}
ReadonlySpan<MP3::Tables::ScaleFactorBand> MP3LoaderPlugin::get_scalefactor_bands(MP3::Granule const& granule, int samplerate)
{
switch (granule.block_type) {
case MP3::BlockType::Short:
switch (samplerate) {
case 32000:
return granule.mixed_block_flag ? MP3::Tables::ScaleFactorBandMixed32000 : MP3::Tables::ScaleFactorBandShort32000;
case 44100:
return granule.mixed_block_flag ? MP3::Tables::ScaleFactorBandMixed44100 : MP3::Tables::ScaleFactorBandShort44100;
case 48000:
return granule.mixed_block_flag ? MP3::Tables::ScaleFactorBandMixed48000 : MP3::Tables::ScaleFactorBandShort48000;
}
break;
case MP3::BlockType::Normal:
[[fallthrough]];
case MP3::BlockType::Start:
[[fallthrough]];
case MP3::BlockType::End:
switch (samplerate) {
case 32000:
return MP3::Tables::ScaleFactorBandLong32000;
case 44100:
return MP3::Tables::ScaleFactorBandLong44100;
case 48000:
return MP3::Tables::ScaleFactorBandLong48000;
}
}
VERIFY_NOT_REACHED();
}
}
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